15 Oct
2018
15 Oct
'18
12:15 p.m.
Hello,
indeed, I was misled by the Route headers in INVITE, which looked like
inside a dialog, but the parameter in To header is rinstance.
Is there any 18x response?
Cheers,
Daniel
On 15.10.18 16:00, Sergiu Pojoga wrote:
Hi again,
Hmm... I don't see a To-tag in the INVITE, neither there's a 200OK to
provide because the UPDATE was sent out prior to the callee answering
the call.
If there should be a Route header in the UPDATE, it would it indicate
a bug with Asterisk firing off the UPDATE without a pre-set Route
dictated by the Path?
If that's the case, I suppose my options are:
1. reach out to Asterisk to investigate and fix it (unrealistic)
2. store the Route header from the initial INVITE in a AVP and insert
it later if an UPDATE follows. Would that break anything up?
Any other constructive suggestions?
Thanks.
On Mon, Oct 15, 2018 at 2:34 AM Daniel-Constantin Mierla
<miconda(a)gmail.com <mailto:miconda@gmail.com>> wrote:
Hello,
that seems to be a re-INVITE (has To-tag). I would need at least
the initial INVITE and the 200ok, along with the UPDATE request.
If the UPDATE is after the re-INVITE, it is missing the Route
header as in the re-INVITE.
Cheers,
Daniel
On 12.10.18 16:53, Sergiu Pojoga wrote:
--
Daniel-Constantin Mierla -- www.asipto.com
www.twitter.com/miconda -- www.linkedin.com/in/miconda
Kamailio World Conference -- www.kamailioworld.com
Kamailio Advanced Training, Nov 12-14, 2018, in Berlin -- www.asipto.com
Hi Daniel,
Certainly, below find the initial INVITE and the subsequent
UPDATE, as received by Kamailio(a)65.xx.xx.167
<mailto:Kamailio@65.xx.xx.167>. If those aren't sufficient, let
me know and if it's ok with you, I'll send the full pcap in private.
The dilemma in my mind is whether the UPDATE should have a
pre-set Route header, similar to how the INVITE has.
2018/10/11 12:34:57.339306 65.xx.xx.172:5060 ->
65.xx.xx.167:5060
INVITE sip:238@65.xx.xx.161:64877;rinstance=8a315091627cc10b
SIP/2.0
Via: SIP/2.0/UDP 65.xx.xx.172:5060;branch=z9hG4bK694382a1
Max-Forwards: 70
Route:
<sip:65.xx.xx.167;lr;received=sip:65.xx.xx.161:64877;r2=on>,<sip:xx.xx.xx.167:5070;lr;received=sip:65.xx.xx.161:64877;r2=on>
From: "Robert" <sip:226@mypbx.net
<mailto:sip%3A226@mypbx.net>>;tag=as0ecef1c4
To: <sip:238@65.xx.xx.161:64877;rinstance=8a315091627cc10b>
Contact: <sip:226@65.xx.xx.172:5060>
Call-ID: 1e82197b42f0173b25e70759753d4210(a)mypbx.net
<mailto:1e82197b42f0173b25e70759753d4210@mypbx.net>
CSeq: 102 INVITE
Supported: replaces, timer, path
Content-Type: application/sdp
Content-Length: 386
2018/10/11 12:35:06.096457 65.xx.xx.172:5060 ->
65.xx.xx.167:5060
UPDATE sip:238@10.17.0.35:64877;alias=65.xx.xx.161~64877~1
SIP/2.0
Via: SIP/2.0/UDP 65.xx.xx.172:5060;branch=z9hG4bK34fab05c
Max-Forwards: 70
From: "Robert" <sip:226@mypbx.net
<mailto:sip%3A226@mypbx.net>>;tag=as0ecef1c4
To:
<sip:238@65.xx.xx.161:64877;rinstance=8a315091627cc10b>;tag=6467b07f
Contact: <sip:226@65.xx.xx.172:5060>
Call-ID: 1e82197b42f0173b25e70759753d4210(a)mypbx.net
<mailto:1e82197b42f0173b25e70759753d4210@mypbx.net>
CSeq: 103 UPDATE
Content-Length: 0
Much obliged.
On Fri, Oct 12, 2018 at 9:38 AM Daniel-Constantin Mierla
<miconda(a)gmail.com <mailto:miconda@gmail.com>> wrote:
Hello,
you hve to provide the sip traffic for this case, the
screenshot doesn't show the sip headers used for routing in
this case, therefore grab the sip traffic for all sip
messages in such scenarion, either ngrep output or pcap file,
and send it over to see if some headers are missing or not
set properly.
Cheers,
Daniel
On 11.10.18 21:03, Sergiu Pojoga wrote:
--
Daniel-Constantin Mierla -- www.asipto.com <http://www.asipto.com>
www.twitter.com/miconda <http://www.twitter.com/miconda> --
www.linkedin.com/in/miconda <http://www.linkedin.com/in/miconda>
Kamailio World Conference -- www.kamailioworld.com
<http://www.kamailioworld.com>
Kamailio Advanced Training, Nov 12-14, 2018, in Berlin -- www.asipto.com
<http://www.asipto.com>
Hi ppl,
I have this problem with call transfer, may be someone can help.
The phone to the far right is registered with the Registrar
to the far left using two PATH headers (trespassing two
proxy ports, 5070 then 5060).
As you can see in the graph below, after receiving the
UPDATE request, Kamailio relays it further from port 5060, I
expect it to be from 5070 just like the dialog forming
INVITE and the CANCEL afterwards.
image.png
The UPDATE has a to-tag, but unlike the original INVITE - it
has no Route header!???
route[*WITHINDLG*] {
if (!has_totag()) return;
if (loose_route()) {
route(DLGURI);
if (is_method("BYE")) {
...
}
else if ( is_method("ACK") ) {
route(NATMANAGE);
}
else if ( is_method("NOTIFY") ) {
record_route();
}
route(RELAY);
exit;
}
if ( is_method("ACK") ) {
...
}
# handle UPDATE method for in-dialog requests
if (is_method("*UPDATE*")) {
route(DLGURI);
record_route();
route(RELAY);
}
}
Thanks in advance.
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--
Daniel-Constantin Mierla -- www.asipto.com <http://www.asipto.com>
www.twitter.com/miconda <http://www.twitter.com/miconda> --
www.linkedin.com/in/miconda <http://www.linkedin.com/in/miconda>
Kamailio World Conference -- www.kamailioworld.com
<http://www.kamailioworld.com>
Kamailio Advanced Training, Nov 12-14, 2018, in Berlin -- www.asipto.com
<http://www.asipto.com>